AuroraOpenALSoft/Alc/ALu.c

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/**
* OpenAL cross platform audio library
* Copyright (C) 1999-2007 by authors.
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <math.h>
#include <stdlib.h>
#include <string.h>
#include <ctype.h>
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#include <assert.h>
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#include "alMain.h"
#include "AL/al.h"
#include "AL/alc.h"
#include "alSource.h"
#include "alBuffer.h"
#include "alListener.h"
#include "alAuxEffectSlot.h"
#include "alu.h"
#include "bs2b.h"
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/* Cone scalar */
ALfloat ConeScale = 0.5f;
/* Localized Z scalar for mono sources */
ALfloat ZScale = 1.0f;
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static __inline ALvoid aluCrossproduct(const ALfloat *inVector1, const ALfloat *inVector2, ALfloat *outVector)
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{
outVector[0] = inVector1[1]*inVector2[2] - inVector1[2]*inVector2[1];
outVector[1] = inVector1[2]*inVector2[0] - inVector1[0]*inVector2[2];
outVector[2] = inVector1[0]*inVector2[1] - inVector1[1]*inVector2[0];
}
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static __inline ALfloat aluDotproduct(const ALfloat *inVector1, const ALfloat *inVector2)
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{
return inVector1[0]*inVector2[0] + inVector1[1]*inVector2[1] +
inVector1[2]*inVector2[2];
}
static __inline ALfloat aluNormalize(ALfloat *inVector)
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{
ALfloat length, inverse_length;
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length = aluSqrt(aluDotproduct(inVector, inVector));
if(length > 0.0f)
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{
inverse_length = 1.0f/length;
inVector[0] *= inverse_length;
inVector[1] *= inverse_length;
inVector[2] *= inverse_length;
}
return length;
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}
static __inline ALvoid aluMatrixVector(ALfloat *vector,ALfloat w,ALfloat matrix[4][4])
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{
ALfloat temp[4] = {
vector[0], vector[1], vector[2], w
};
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vector[0] = temp[0]*matrix[0][0] + temp[1]*matrix[1][0] + temp[2]*matrix[2][0] + temp[3]*matrix[3][0];
vector[1] = temp[0]*matrix[0][1] + temp[1]*matrix[1][1] + temp[2]*matrix[2][1] + temp[3]*matrix[3][1];
vector[2] = temp[0]*matrix[0][2] + temp[1]*matrix[1][2] + temp[2]*matrix[2][2] + temp[3]*matrix[3][2];
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}
ALvoid CalcNonAttnSourceParams(ALsource *ALSource, const ALCcontext *ALContext)
{
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static const ALfloat angles_Mono[1] = { 0.0f };
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static const ALfloat angles_Stereo[2] = { -30.0f, 30.0f };
static const ALfloat angles_Rear[2] = { -150.0f, 150.0f };
static const ALfloat angles_Quad[4] = { -45.0f, 45.0f, -135.0f, 135.0f };
static const ALfloat angles_X51[6] = { -30.0f, 30.0f, 0.0f, 0.0f,
-110.0f, 110.0f };
static const ALfloat angles_X61[7] = { -30.0f, 30.0f, 0.0f, 0.0f,
180.0f, -90.0f, 90.0f };
static const ALfloat angles_X71[8] = { -30.0f, 30.0f, 0.0f, 0.0f,
-110.0f, 110.0f, -90.0f, 90.0f };
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static const enum Channel chans_Mono[1] = { FRONT_CENTER };
static const enum Channel chans_Stereo[2] = { FRONT_LEFT, FRONT_RIGHT };
static const enum Channel chans_Rear[2] = { BACK_LEFT, BACK_RIGHT };
static const enum Channel chans_Quad[4] = { FRONT_LEFT, FRONT_RIGHT,
BACK_LEFT, BACK_RIGHT };
static const enum Channel chans_X51[6] = { FRONT_LEFT, FRONT_RIGHT,
FRONT_CENTER, LFE,
BACK_LEFT, BACK_RIGHT };
static const enum Channel chans_X61[7] = { FRONT_LEFT, FRONT_RIGHT,
FRONT_CENTER, LFE, BACK_CENTER,
SIDE_LEFT, SIDE_RIGHT };
static const enum Channel chans_X71[8] = { FRONT_LEFT, FRONT_RIGHT,
FRONT_CENTER, LFE,
BACK_LEFT, BACK_RIGHT,
SIDE_LEFT, SIDE_RIGHT };
ALCdevice *Device = ALContext->Device;
ALfloat SourceVolume,ListenerGain,MinVolume,MaxVolume;
ALbufferlistitem *BufferListItem;
enum DevFmtChannels DevChans;
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enum FmtChannels Channels;
ALfloat (*SrcMatrix)[MAXCHANNELS];
ALfloat DryGain, DryGainHF;
ALfloat WetGain[MAX_SENDS];
ALfloat WetGainHF[MAX_SENDS];
ALint NumSends, Frequency;
const ALfloat *SpeakerGain;
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const ALfloat *angles = NULL;
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const enum Channel *chans = NULL;
enum Resampler Resampler;
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ALint num_channels = 0;
ALboolean VirtualChannels;
ALfloat Pitch;
ALfloat cw;
ALuint pos;
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ALint i, c;
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/* Get device properties */
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DevChans = Device->FmtChans;
NumSends = Device->NumAuxSends;
Frequency = Device->Frequency;
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/* Get listener properties */
ListenerGain = ALContext->Listener.Gain;
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/* Get source properties */
SourceVolume = ALSource->flGain;
MinVolume = ALSource->flMinGain;
MaxVolume = ALSource->flMaxGain;
Pitch = ALSource->flPitch;
Resampler = ALSource->Resampler;
VirtualChannels = ALSource->VirtualChannels;
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/* Calculate the stepping value */
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Channels = FmtMono;
BufferListItem = ALSource->queue;
while(BufferListItem != NULL)
{
ALbuffer *ALBuffer;
if((ALBuffer=BufferListItem->buffer) != NULL)
{
ALint maxstep = STACK_DATA_SIZE / ALSource->NumChannels /
ALSource->SampleSize;
maxstep -= ResamplerPadding[Resampler] +
ResamplerPrePadding[Resampler] + 1;
maxstep = mini(maxstep, INT_MAX>>FRACTIONBITS);
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Pitch = Pitch * ALBuffer->Frequency / Frequency;
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if(Pitch > (ALfloat)maxstep)
ALSource->Params.Step = maxstep<<FRACTIONBITS;
else
{
ALSource->Params.Step = fastf2i(Pitch*FRACTIONONE);
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if(ALSource->Params.Step == 0)
ALSource->Params.Step = 1;
}
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Channels = ALBuffer->FmtChannels;
break;
}
BufferListItem = BufferListItem->next;
}
if(VirtualChannels && Device->Hrtf)
ALSource->Params.DoMix = SelectHrtfMixer((ALSource->Params.Step==FRACTIONONE) ?
POINT_RESAMPLER : Resampler);
else
ALSource->Params.DoMix = SelectMixer((ALSource->Params.Step==FRACTIONONE) ?
POINT_RESAMPLER : Resampler);
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/* Calculate gains */
DryGain = clampf(SourceVolume, MinVolume, MaxVolume);
DryGain *= ALSource->DirectGain;
DryGainHF = ALSource->DirectGainHF;
for(i = 0;i < NumSends;i++)
{
WetGain[i] = clampf(SourceVolume, MinVolume, MaxVolume);
WetGain[i] *= ALSource->Send[i].WetGain;
WetGainHF[i] = ALSource->Send[i].WetGainHF;
}
SrcMatrix = ALSource->Params.DryGains;
for(i = 0;i < MAXCHANNELS;i++)
{
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for(c = 0;c < MAXCHANNELS;c++)
SrcMatrix[i][c] = 0.0f;
}
switch(Channels)
{
case FmtMono:
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angles = angles_Mono;
chans = chans_Mono;
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num_channels = 1;
break;
case FmtStereo:
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if(VirtualChannels && (Device->Flags&DEVICE_DUPLICATE_STEREO))
{
DryGain *= aluSqrt(2.0f/4.0f);
for(c = 0;c < 2;c++)
{
pos = aluCart2LUTpos(aluCos(F_PI/180.0f * angles_Rear[c]),
aluSin(F_PI/180.0f * angles_Rear[c]));
SpeakerGain = Device->PanningLUT[pos];
for(i = 0;i < (ALint)Device->NumChan;i++)
{
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enum Channel chan = Device->Speaker2Chan[i];
SrcMatrix[c][chan] += DryGain * ListenerGain *
SpeakerGain[chan];
}
}
}
angles = angles_Stereo;
chans = chans_Stereo;
num_channels = 2;
break;
case FmtRear:
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angles = angles_Rear;
chans = chans_Rear;
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num_channels = 2;
break;
case FmtQuad:
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angles = angles_Quad;
chans = chans_Quad;
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num_channels = 4;
break;
case FmtX51:
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angles = angles_X51;
chans = chans_X51;
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num_channels = 6;
break;
case FmtX61:
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angles = angles_X61;
chans = chans_X61;
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num_channels = 7;
break;
case FmtX71:
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angles = angles_X71;
chans = chans_X71;
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num_channels = 8;
break;
}
if(VirtualChannels == AL_FALSE)
{
for(c = 0;c < num_channels;c++)
SrcMatrix[c][chans[c]] += DryGain * ListenerGain;
}
else if(Device->Hrtf)
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{
for(c = 0;c < num_channels;c++)
{
if(chans[c] == LFE)
{
/* Skip LFE */
ALSource->Params.HrtfDelay[c][0] = 0;
ALSource->Params.HrtfDelay[c][1] = 0;
for(i = 0;i < HRIR_LENGTH;i++)
{
ALSource->Params.HrtfCoeffs[c][i][0] = 0.0f;
ALSource->Params.HrtfCoeffs[c][i][1] = 0.0f;
}
}
else
{
/* Get the static HRIR coefficients and delays for this
* channel. */
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GetLerpedHrtfCoeffs(Device->Hrtf,
0.0f, F_PI/180.0f * angles[c],
DryGain*ListenerGain,
ALSource->Params.HrtfCoeffs[c],
ALSource->Params.HrtfDelay[c]);
}
ALSource->HrtfCounter = 0;
}
}
else
{
for(c = 0;c < num_channels;c++)
{
if(chans[c] == LFE) /* Special-case LFE */
{
SrcMatrix[c][LFE] += DryGain * ListenerGain;
continue;
}
pos = aluCart2LUTpos(aluCos(F_PI/180.0f * angles[c]),
aluSin(F_PI/180.0f * angles[c]));
SpeakerGain = Device->PanningLUT[pos];
for(i = 0;i < (ALint)Device->NumChan;i++)
{
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enum Channel chan = Device->Speaker2Chan[i];
SrcMatrix[c][chan] += DryGain * ListenerGain *
SpeakerGain[chan];
}
}
}
for(i = 0;i < NumSends;i++)
{
ALSource->Params.Send[i].Slot = ALSource->Send[i].Slot;
ALSource->Params.Send[i].WetGain = WetGain[i] * ListenerGain;
}
/* Update filter coefficients. Calculations based on the I3DL2
* spec. */
cw = aluCos(F_PI*2.0f * LOWPASSFREQREF / Frequency);
/* We use two chained one-pole filters, so we need to take the
* square root of the squared gain, which is the same as the base
* gain. */
ALSource->Params.iirFilter.coeff = lpCoeffCalc(DryGainHF, cw);
for(i = 0;i < NumSends;i++)
{
/* We use a one-pole filter, so we need to take the squared gain */
ALfloat a = lpCoeffCalc(WetGainHF[i]*WetGainHF[i], cw);
ALSource->Params.Send[i].iirFilter.coeff = a;
}
}
ALvoid CalcSourceParams(ALsource *ALSource, const ALCcontext *ALContext)
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{
const ALCdevice *Device = ALContext->Device;
ALfloat InnerAngle,OuterAngle,Angle,Distance,ClampedDist;
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ALfloat Direction[3],Position[3],SourceToListener[3];
ALfloat Velocity[3],ListenerVel[3];
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ALfloat MinVolume,MaxVolume,MinDist,MaxDist,Rolloff;
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ALfloat ConeVolume,ConeHF,SourceVolume,ListenerGain;
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ALfloat DopplerFactor, DopplerVelocity, SpeedOfSound;
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ALfloat AirAbsorptionFactor;
ALfloat RoomAirAbsorption[MAX_SENDS];
ALbufferlistitem *BufferListItem;
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ALfloat Attenuation, EffectiveDist;
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ALfloat RoomAttenuation[MAX_SENDS];
ALfloat MetersPerUnit;
ALfloat RoomRolloffBase;
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ALfloat RoomRolloff[MAX_SENDS];
ALfloat DecayDistance[MAX_SENDS];
ALfloat DryGain;
ALfloat DryGainHF;
ALboolean DryGainHFAuto;
ALfloat WetGain[MAX_SENDS];
ALfloat WetGainHF[MAX_SENDS];
ALboolean WetGainAuto;
ALboolean WetGainHFAuto;
enum Resampler Resampler;
ALfloat Pitch;
ALuint Frequency;
ALint NumSends;
ALfloat cw;
ALint i;
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DryGainHF = 1.0f;
for(i = 0;i < MAX_SENDS;i++)
WetGainHF[i] = 1.0f;
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//Get context properties
DopplerFactor = ALContext->DopplerFactor * ALSource->DopplerFactor;
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DopplerVelocity = ALContext->DopplerVelocity;
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SpeedOfSound = ALContext->flSpeedOfSound;
NumSends = Device->NumAuxSends;
Frequency = Device->Frequency;
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//Get listener properties
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ListenerGain = ALContext->Listener.Gain;
MetersPerUnit = ALContext->Listener.MetersPerUnit;
ListenerVel[0] = ALContext->Listener.Velocity[0];
ListenerVel[1] = ALContext->Listener.Velocity[1];
ListenerVel[2] = ALContext->Listener.Velocity[2];
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//Get source properties
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SourceVolume = ALSource->flGain;
MinVolume = ALSource->flMinGain;
MaxVolume = ALSource->flMaxGain;
Pitch = ALSource->flPitch;
Resampler = ALSource->Resampler;
Position[0] = ALSource->vPosition[0];
Position[1] = ALSource->vPosition[1];
Position[2] = ALSource->vPosition[2];
Direction[0] = ALSource->vOrientation[0];
Direction[1] = ALSource->vOrientation[1];
Direction[2] = ALSource->vOrientation[2];
Velocity[0] = ALSource->vVelocity[0];
Velocity[1] = ALSource->vVelocity[1];
Velocity[2] = ALSource->vVelocity[2];
MinDist = ALSource->flRefDistance;
MaxDist = ALSource->flMaxDistance;
Rolloff = ALSource->flRollOffFactor;
InnerAngle = ALSource->flInnerAngle * ConeScale;
OuterAngle = ALSource->flOuterAngle * ConeScale;
AirAbsorptionFactor = ALSource->AirAbsorptionFactor;
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DryGainHFAuto = ALSource->DryGainHFAuto;
WetGainAuto = ALSource->WetGainAuto;
WetGainHFAuto = ALSource->WetGainHFAuto;
RoomRolloffBase = ALSource->RoomRolloffFactor;
for(i = 0;i < NumSends;i++)
{
ALeffectslot *Slot = ALSource->Send[i].Slot;
if(!Slot || Slot->effect.type == AL_EFFECT_NULL)
{
RoomRolloff[i] = 0.0f;
DecayDistance[i] = 0.0f;
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RoomAirAbsorption[i] = 1.0f;
}
else if(Slot->AuxSendAuto)
{
RoomRolloff[i] = RoomRolloffBase;
if(IsReverbEffect(Slot->effect.type))
{
RoomRolloff[i] += Slot->effect.Reverb.RoomRolloffFactor;
DecayDistance[i] = Slot->effect.Reverb.DecayTime *
SPEEDOFSOUNDMETRESPERSEC;
RoomAirAbsorption[i] = Slot->effect.Reverb.AirAbsorptionGainHF;
}
else
{
DecayDistance[i] = 0.0f;
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RoomAirAbsorption[i] = 1.0f;
}
}
else
{
/* If the slot's auxiliary send auto is off, the data sent to the
* effect slot is the same as the dry path, sans filter effects */
RoomRolloff[i] = Rolloff;
DecayDistance[i] = 0.0f;
RoomAirAbsorption[i] = AIRABSORBGAINHF;
}
ALSource->Params.Send[i].Slot = Slot;
}
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//1. Translate Listener to origin (convert to head relative)
if(ALSource->bHeadRelative == AL_FALSE)
{
ALfloat U[3],V[3],N[3];
ALfloat Matrix[4][4];
// Build transform matrix
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N[0] = ALContext->Listener.Forward[0]; // At-vector
N[1] = ALContext->Listener.Forward[1];
N[2] = ALContext->Listener.Forward[2];
V[0] = ALContext->Listener.Up[0]; // Up-vector
V[1] = ALContext->Listener.Up[1];
V[2] = ALContext->Listener.Up[2];
if(aluNormalize(N) > 0.0f && aluNormalize(V) > 0.0f)
{
/* Build and normalize right-vector */
aluCrossproduct(N, V, U);
if(aluNormalize(U) > 0.0f)
{
Matrix[0][0]=U[0]; Matrix[0][1]=V[0]; Matrix[0][2]=-N[0]; Matrix[0][3]=0.0f;
Matrix[1][0]=U[1]; Matrix[1][1]=V[1]; Matrix[1][2]=-N[1]; Matrix[1][3]=0.0f;
Matrix[2][0]=U[2]; Matrix[2][1]=V[2]; Matrix[2][2]=-N[2]; Matrix[2][3]=0.0f;
Matrix[3][0]=0.0f; Matrix[3][1]=0.0f; Matrix[3][2]= 0.0f; Matrix[3][3]=1.0f;
/* Translate position */
Position[0] -= ALContext->Listener.Position[0];
Position[1] -= ALContext->Listener.Position[1];
Position[2] -= ALContext->Listener.Position[2];
/* Transform source vectors into listener space */
aluMatrixVector(Position, 1.0f, Matrix);
aluMatrixVector(Direction, 0.0f, Matrix);
aluMatrixVector(Velocity, 0.0f, Matrix);
}
}
}
else
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{
ListenerVel[0] = 0.0f;
ListenerVel[1] = 0.0f;
ListenerVel[2] = 0.0f;
}
SourceToListener[0] = -Position[0];
SourceToListener[1] = -Position[1];
SourceToListener[2] = -Position[2];
aluNormalize(SourceToListener);
aluNormalize(Direction);
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//2. Calculate distance attenuation
Distance = aluSqrt(aluDotproduct(Position, Position));
ClampedDist = Distance;
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Attenuation = 1.0f;
for(i = 0;i < NumSends;i++)
RoomAttenuation[i] = 1.0f;
switch(ALContext->SourceDistanceModel ? ALSource->DistanceModel :
ALContext->DistanceModel)
{
case InverseDistanceClamped:
ClampedDist = clampf(ClampedDist, MinDist, MaxDist);
if(MaxDist < MinDist)
break;
//fall-through
case InverseDistance:
if(MinDist > 0.0f)
{
if((MinDist + (Rolloff * (ClampedDist - MinDist))) > 0.0f)
Attenuation = MinDist / (MinDist + (Rolloff * (ClampedDist - MinDist)));
for(i = 0;i < NumSends;i++)
{
if((MinDist + (RoomRolloff[i] * (ClampedDist - MinDist))) > 0.0f)
RoomAttenuation[i] = MinDist / (MinDist + (RoomRolloff[i] * (ClampedDist - MinDist)));
}
}
break;
case LinearDistanceClamped:
ClampedDist = clampf(ClampedDist, MinDist, MaxDist);
if(MaxDist < MinDist)
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break;
//fall-through
case LinearDistance:
if(MaxDist != MinDist)
{
Attenuation = 1.0f - (Rolloff*(ClampedDist-MinDist)/(MaxDist - MinDist));
Attenuation = maxf(Attenuation, 0.0f);
for(i = 0;i < NumSends;i++)
{
RoomAttenuation[i] = 1.0f - (RoomRolloff[i]*(ClampedDist-MinDist)/(MaxDist - MinDist));
RoomAttenuation[i] = maxf(RoomAttenuation[i], 0.0f);
}
}
break;
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case ExponentDistanceClamped:
ClampedDist = clampf(ClampedDist, MinDist, MaxDist);
if(MaxDist < MinDist)
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break;
//fall-through
case ExponentDistance:
if(ClampedDist > 0.0f && MinDist > 0.0f)
{
Attenuation = aluPow(ClampedDist/MinDist, -Rolloff);
for(i = 0;i < NumSends;i++)
RoomAttenuation[i] = aluPow(ClampedDist/MinDist, -RoomRolloff[i]);
}
break;
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case DisableDistance:
break;
}
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// Source Gain + Attenuation
DryGain = SourceVolume * Attenuation;
for(i = 0;i < NumSends;i++)
WetGain[i] = SourceVolume * RoomAttenuation[i];
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// Distance-based air absorption
EffectiveDist = 0.0f;
if(MinDist > 0.0f && Attenuation < 1.0f)
EffectiveDist = (MinDist/Attenuation - MinDist)*MetersPerUnit;
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if(AirAbsorptionFactor > 0.0f && EffectiveDist > 0.0f)
{
DryGainHF *= aluPow(AIRABSORBGAINHF, AirAbsorptionFactor*EffectiveDist);
for(i = 0;i < NumSends;i++)
WetGainHF[i] *= aluPow(RoomAirAbsorption[i],
AirAbsorptionFactor*EffectiveDist);
}
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if(WetGainAuto)
{
/* Apply a decay-time transformation to the wet path, based on the
* attenuation of the dry path.
*
* Using the approximate (effective) source to listener distance, the
* initial decay of the reverb effect is calculated and applied to the
* wet path.
*/
for(i = 0;i < NumSends;i++)
{
if(DecayDistance[i] > 0.0f)
WetGain[i] *= aluPow(0.001f /* -60dB */,
EffectiveDist / DecayDistance[i]);
}
}
/* Calculate directional soundcones */
Angle = aluAcos(aluDotproduct(Direction,SourceToListener)) * (180.0f/F_PI);
if(Angle >= InnerAngle && Angle <= OuterAngle)
{
ALfloat scale = (Angle-InnerAngle) / (OuterAngle-InnerAngle);
ConeVolume = lerp(1.0f, ALSource->flOuterGain, scale);
ConeHF = lerp(1.0f, ALSource->OuterGainHF, scale);
}
else if(Angle > OuterAngle)
{
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ConeVolume = ALSource->flOuterGain;
ConeHF = ALSource->OuterGainHF;
}
else
{
ConeVolume = 1.0f;
ConeHF = 1.0f;
}
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DryGain *= ConeVolume;
if(WetGainAuto)
{
for(i = 0;i < NumSends;i++)
WetGain[i] *= ConeVolume;
}
if(DryGainHFAuto)
DryGainHF *= ConeHF;
if(WetGainHFAuto)
{
for(i = 0;i < NumSends;i++)
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WetGainHF[i] *= ConeHF;
}
// Clamp to Min/Max Gain
DryGain = clampf(DryGain, MinVolume, MaxVolume);
for(i = 0;i < NumSends;i++)
WetGain[i] = clampf(WetGain[i], MinVolume, MaxVolume);
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// Apply filter gains and filters
DryGain *= ALSource->DirectGain * ListenerGain;
DryGainHF *= ALSource->DirectGainHF;
for(i = 0;i < NumSends;i++)
{
WetGain[i] *= ALSource->Send[i].WetGain * ListenerGain;
WetGainHF[i] *= ALSource->Send[i].WetGainHF;
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}
// Calculate Velocity
if(DopplerFactor != 0.0f)
{
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ALfloat VSS, VLS;
ALfloat MaxVelocity = (SpeedOfSound*DopplerVelocity) /
DopplerFactor;
VSS = aluDotproduct(Velocity, SourceToListener);
if(VSS >= MaxVelocity)
VSS = (MaxVelocity - 1.0f);
else if(VSS <= -MaxVelocity)
VSS = -MaxVelocity + 1.0f;
VLS = aluDotproduct(ListenerVel, SourceToListener);
if(VLS >= MaxVelocity)
VLS = (MaxVelocity - 1.0f);
else if(VLS <= -MaxVelocity)
VLS = -MaxVelocity + 1.0f;
Pitch *= ((SpeedOfSound*DopplerVelocity) - (DopplerFactor*VLS)) /
((SpeedOfSound*DopplerVelocity) - (DopplerFactor*VSS));
}
BufferListItem = ALSource->queue;
while(BufferListItem != NULL)
{
ALbuffer *ALBuffer;
if((ALBuffer=BufferListItem->buffer) != NULL)
{
ALint maxstep = STACK_DATA_SIZE / ALSource->NumChannels /
ALSource->SampleSize;
maxstep -= ResamplerPadding[Resampler] +
ResamplerPrePadding[Resampler] + 1;
maxstep = mini(maxstep, INT_MAX>>FRACTIONBITS);
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Pitch = Pitch * ALBuffer->Frequency / Frequency;
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if(Pitch > (ALfloat)maxstep)
ALSource->Params.Step = maxstep<<FRACTIONBITS;
else
{
ALSource->Params.Step = fastf2i(Pitch*FRACTIONONE);
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if(ALSource->Params.Step == 0)
ALSource->Params.Step = 1;
}
break;
}
BufferListItem = BufferListItem->next;
}
if(Device->Hrtf)
ALSource->Params.DoMix = SelectHrtfMixer((ALSource->Params.Step==FRACTIONONE) ?
POINT_RESAMPLER : Resampler);
else
ALSource->Params.DoMix = SelectMixer((ALSource->Params.Step==FRACTIONONE) ?
POINT_RESAMPLER : Resampler);
if(Device->Hrtf)
{
// Use a binaural HRTF algorithm for stereo headphone playback
ALfloat delta, ev = 0.0f, az = 0.0f;
if(Distance > 0.0f)
{
ALfloat invlen = 1.0f/Distance;
Position[0] *= invlen;
Position[1] *= invlen;
Position[2] *= invlen;
// Calculate elevation and azimuth only when the source is not at
// the listener. This prevents +0 and -0 Z from producing
// inconsistent panning.
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ev = aluAsin(Position[1]);
az = aluAtan2(Position[0], -Position[2]*ZScale);
}
// Check to see if the HRIR is already moving.
if(ALSource->HrtfMoving)
{
// Calculate the normalized HRTF transition factor (delta).
delta = CalcHrtfDelta(ALSource->Params.HrtfGain, DryGain,
ALSource->Params.HrtfDir, Position);
// If the delta is large enough, get the moving HRIR target
// coefficients, target delays, steppping values, and counter.
if(delta > 0.001f)
{
ALSource->HrtfCounter = GetMovingHrtfCoeffs(Device->Hrtf,
ev, az, DryGain, delta,
ALSource->HrtfCounter,
ALSource->Params.HrtfCoeffs[0],
ALSource->Params.HrtfDelay[0],
ALSource->Params.HrtfCoeffStep,
ALSource->Params.HrtfDelayStep);
ALSource->Params.HrtfGain = DryGain;
ALSource->Params.HrtfDir[0] = Position[0];
ALSource->Params.HrtfDir[1] = Position[1];
ALSource->Params.HrtfDir[2] = Position[2];
}
}
else
{
// Get the initial (static) HRIR coefficients and delays.
GetLerpedHrtfCoeffs(Device->Hrtf, ev, az, DryGain,
ALSource->Params.HrtfCoeffs[0],
ALSource->Params.HrtfDelay[0]);
ALSource->HrtfCounter = 0;
ALSource->Params.HrtfGain = DryGain;
ALSource->Params.HrtfDir[0] = Position[0];
ALSource->Params.HrtfDir[1] = Position[1];
ALSource->Params.HrtfDir[2] = Position[2];
}
}
else
{
// Use energy-preserving panning algorithm for multi-speaker playback
ALfloat DirGain, AmbientGain;
const ALfloat *SpeakerGain;
ALfloat length;
ALint pos;
length = maxf(Distance, MinDist);
if(length > 0.0f)
{
ALfloat invlen = 1.0f/length;
Position[0] *= invlen;
Position[1] *= invlen;
Position[2] *= invlen;
}
pos = aluCart2LUTpos(-Position[2]*ZScale, Position[0]);
SpeakerGain = Device->PanningLUT[pos];
DirGain = aluSqrt(Position[0]*Position[0] + Position[2]*Position[2]);
// elevation adjustment for directional gain. this sucks, but
// has low complexity
2011-09-24 05:50:33 +00:00
AmbientGain = aluSqrt(1.0f/Device->NumChan);
for(i = 0;i < MAXCHANNELS;i++)
{
ALuint i2;
for(i2 = 0;i2 < MAXCHANNELS;i2++)
ALSource->Params.DryGains[i][i2] = 0.0f;
}
for(i = 0;i < (ALint)Device->NumChan;i++)
{
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enum Channel chan = Device->Speaker2Chan[i];
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ALfloat gain = lerp(AmbientGain, SpeakerGain[chan], DirGain);
ALSource->Params.DryGains[0][chan] = DryGain * gain;
}
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}
for(i = 0;i < NumSends;i++)
ALSource->Params.Send[i].WetGain = WetGain[i];
/* Update filter coefficients. */
cw = aluCos(F_PI*2.0f * LOWPASSFREQREF / Frequency);
ALSource->Params.iirFilter.coeff = lpCoeffCalc(DryGainHF, cw);
for(i = 0;i < NumSends;i++)
{
ALfloat a = lpCoeffCalc(WetGainHF[i]*WetGainHF[i], cw);
ALSource->Params.Send[i].iirFilter.coeff = a;
}
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}
static __inline ALfloat aluF2F(ALfloat val)
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{ return val; }
static __inline ALshort aluF2S(ALfloat val)
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{
if(val > 1.0f) return 32767;
if(val < -1.0f) return -32768;
return fastf2i(val*32767.0f);
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}
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static __inline ALushort aluF2US(ALfloat val)
{ return aluF2S(val)+32768; }
static __inline ALbyte aluF2B(ALfloat val)
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{ return aluF2S(val)>>8; }
static __inline ALubyte aluF2UB(ALfloat val)
{ return aluF2US(val)>>8; }
2010-11-21 10:51:18 +00:00
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#define DECL_TEMPLATE(T, N, func) \
static void Write_##T##_##N(ALCdevice *device, T *RESTRICT buffer, \
ALuint SamplesToDo) \
{ \
ALfloat (*RESTRICT DryBuffer)[MAXCHANNELS] = device->DryBuffer; \
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const enum Channel *ChanMap = device->DevChannels; \
ALuint i, j; \
\
for(i = 0;i < SamplesToDo;i++) \
{ \
for(j = 0;j < N;j++) \
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*(buffer++) = func(DryBuffer[i][ChanMap[j]]); \
} \
}
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DECL_TEMPLATE(ALfloat, 1, aluF2F)
DECL_TEMPLATE(ALfloat, 4, aluF2F)
DECL_TEMPLATE(ALfloat, 6, aluF2F)
DECL_TEMPLATE(ALfloat, 7, aluF2F)
DECL_TEMPLATE(ALfloat, 8, aluF2F)
DECL_TEMPLATE(ALushort, 1, aluF2US)
DECL_TEMPLATE(ALushort, 4, aluF2US)
DECL_TEMPLATE(ALushort, 6, aluF2US)
DECL_TEMPLATE(ALushort, 7, aluF2US)
DECL_TEMPLATE(ALushort, 8, aluF2US)
DECL_TEMPLATE(ALshort, 1, aluF2S)
DECL_TEMPLATE(ALshort, 4, aluF2S)
DECL_TEMPLATE(ALshort, 6, aluF2S)
DECL_TEMPLATE(ALshort, 7, aluF2S)
DECL_TEMPLATE(ALshort, 8, aluF2S)
DECL_TEMPLATE(ALubyte, 1, aluF2UB)
DECL_TEMPLATE(ALubyte, 4, aluF2UB)
DECL_TEMPLATE(ALubyte, 6, aluF2UB)
DECL_TEMPLATE(ALubyte, 7, aluF2UB)
DECL_TEMPLATE(ALubyte, 8, aluF2UB)
DECL_TEMPLATE(ALbyte, 1, aluF2B)
DECL_TEMPLATE(ALbyte, 4, aluF2B)
DECL_TEMPLATE(ALbyte, 6, aluF2B)
DECL_TEMPLATE(ALbyte, 7, aluF2B)
DECL_TEMPLATE(ALbyte, 8, aluF2B)
#undef DECL_TEMPLATE
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#define DECL_TEMPLATE(T, N, func) \
static void Write_##T##_##N(ALCdevice *device, T *RESTRICT buffer, \
ALuint SamplesToDo) \
{ \
2011-05-07 11:32:53 +00:00
ALfloat (*RESTRICT DryBuffer)[MAXCHANNELS] = device->DryBuffer; \
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const enum Channel *ChanMap = device->DevChannels; \
ALuint i, j; \
\
if(device->Bs2b) \
{ \
for(i = 0;i < SamplesToDo;i++) \
{ \
float samples[2]; \
2011-07-17 08:55:25 +00:00
samples[0] = DryBuffer[i][ChanMap[0]]; \
samples[1] = DryBuffer[i][ChanMap[1]]; \
bs2b_cross_feed(device->Bs2b, samples); \
2011-07-17 08:55:25 +00:00
*(buffer++) = func(samples[0]); \
*(buffer++) = func(samples[1]); \
} \
} \
else \
{ \
for(i = 0;i < SamplesToDo;i++) \
{ \
for(j = 0;j < N;j++) \
2011-07-17 08:55:25 +00:00
*(buffer++) = func(DryBuffer[i][ChanMap[j]]); \
} \
} \
}
2011-07-17 08:55:25 +00:00
DECL_TEMPLATE(ALfloat, 2, aluF2F)
DECL_TEMPLATE(ALushort, 2, aluF2US)
DECL_TEMPLATE(ALshort, 2, aluF2S)
DECL_TEMPLATE(ALubyte, 2, aluF2UB)
DECL_TEMPLATE(ALbyte, 2, aluF2B)
#undef DECL_TEMPLATE
2011-05-07 11:31:43 +00:00
#define DECL_TEMPLATE(T) \
static void Write_##T(ALCdevice *device, T *buffer, ALuint SamplesToDo) \
{ \
switch(device->FmtChans) \
{ \
case DevFmtMono: \
2011-07-17 08:55:25 +00:00
Write_##T##_1(device, buffer, SamplesToDo); \
break; \
case DevFmtStereo: \
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Write_##T##_2(device, buffer, SamplesToDo); \
break; \
case DevFmtQuad: \
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Write_##T##_4(device, buffer, SamplesToDo); \
break; \
case DevFmtX51: \
case DevFmtX51Side: \
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Write_##T##_6(device, buffer, SamplesToDo); \
break; \
case DevFmtX61: \
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Write_##T##_7(device, buffer, SamplesToDo); \
break; \
case DevFmtX71: \
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Write_##T##_8(device, buffer, SamplesToDo); \
break; \
} \
}
2011-05-07 11:31:43 +00:00
DECL_TEMPLATE(ALfloat)
DECL_TEMPLATE(ALushort)
DECL_TEMPLATE(ALshort)
DECL_TEMPLATE(ALubyte)
DECL_TEMPLATE(ALbyte)
#undef DECL_TEMPLATE
2010-11-21 10:51:18 +00:00
ALvoid aluMixData(ALCdevice *device, ALvoid *buffer, ALsizei size)
{
ALuint SamplesToDo;
ALeffectslot **slot, **slot_end;
2010-11-21 10:51:18 +00:00
ALsource **src, **src_end;
ALCcontext *ctx;
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int fpuState;
ALuint i, c;
2010-11-21 10:51:18 +00:00
fpuState = SetMixerFPUMode();
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while(size > 0)
{
/* Setup variables */
SamplesToDo = minu(size, BUFFERSIZE);
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/* Clear mixing buffer */
memset(device->DryBuffer, 0, SamplesToDo*MAXCHANNELS*sizeof(ALfloat));
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LockDevice(device);
ctx = device->ContextList;
while(ctx)
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{
ALenum DeferUpdates = ctx->DeferUpdates;
ALenum UpdateSources = AL_FALSE;
if(!DeferUpdates)
UpdateSources = ExchangeInt(&ctx->UpdateSources, AL_FALSE);
src = ctx->ActiveSources;
src_end = src + ctx->ActiveSourceCount;
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while(src != src_end)
{
if((*src)->state != AL_PLAYING)
{
--(ctx->ActiveSourceCount);
2010-11-21 10:51:18 +00:00
*src = *(--src_end);
continue;
}
if(!DeferUpdates && (ExchangeInt(&(*src)->NeedsUpdate, AL_FALSE) ||
UpdateSources))
ALsource_Update(*src, ctx);
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MixSource(*src, device, SamplesToDo);
2010-11-21 10:51:18 +00:00
src++;
}
/* effect slot processing */
slot = ctx->ActiveEffectSlots;
slot_end = slot + ctx->ActiveEffectSlotCount;
while(slot != slot_end)
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{
for(c = 0;c < SamplesToDo;c++)
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{
(*slot)->WetBuffer[c] += (*slot)->ClickRemoval[0];
(*slot)->ClickRemoval[0] -= (*slot)->ClickRemoval[0] * (1.0f/256.0f);
2010-11-21 10:51:18 +00:00
}
(*slot)->ClickRemoval[0] += (*slot)->PendingClicks[0];
(*slot)->PendingClicks[0] = 0.0f;
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if(!DeferUpdates && ExchangeInt(&(*slot)->NeedsUpdate, AL_FALSE))
ALeffectState_Update((*slot)->EffectState, ctx, *slot);
ALeffectState_Process((*slot)->EffectState, SamplesToDo,
(*slot)->WetBuffer, device->DryBuffer);
2010-11-21 10:51:18 +00:00
for(i = 0;i < SamplesToDo;i++)
(*slot)->WetBuffer[i] = 0.0f;
slot++;
2010-11-21 10:51:18 +00:00
}
ctx = ctx->next;
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}
UnlockDevice(device);
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//Post processing loop
if(device->FmtChans == DevFmtMono)
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{
for(i = 0;i < SamplesToDo;i++)
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{
device->DryBuffer[i][FRONT_CENTER] += device->ClickRemoval[FRONT_CENTER];
device->ClickRemoval[FRONT_CENTER] -= device->ClickRemoval[FRONT_CENTER] * (1.0f/256.0f);
2010-11-21 10:51:18 +00:00
}
device->ClickRemoval[FRONT_CENTER] += device->PendingClicks[FRONT_CENTER];
device->PendingClicks[FRONT_CENTER] = 0.0f;
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}
else if(device->FmtChans == DevFmtStereo)
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{
/* Assumes the first two channels are FRONT_LEFT and FRONT_RIGHT */
for(i = 0;i < SamplesToDo;i++)
{
for(c = 0;c < 2;c++)
{
device->DryBuffer[i][c] += device->ClickRemoval[c];
device->ClickRemoval[c] -= device->ClickRemoval[c] * (1.0f/256.0f);
}
}
for(c = 0;c < 2;c++)
{
device->ClickRemoval[c] += device->PendingClicks[c];
device->PendingClicks[c] = 0.0f;
}
}
else
{
for(i = 0;i < SamplesToDo;i++)
{
for(c = 0;c < MAXCHANNELS;c++)
{
device->DryBuffer[i][c] += device->ClickRemoval[c];
device->ClickRemoval[c] -= device->ClickRemoval[c] * (1.0f/256.0f);
}
}
for(c = 0;c < MAXCHANNELS;c++)
{
device->ClickRemoval[c] += device->PendingClicks[c];
device->PendingClicks[c] = 0.0f;
}
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}
if(buffer)
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{
switch(device->FmtType)
{
case DevFmtByte:
Write_ALbyte(device, buffer, SamplesToDo);
break;
case DevFmtUByte:
Write_ALubyte(device, buffer, SamplesToDo);
break;
case DevFmtShort:
Write_ALshort(device, buffer, SamplesToDo);
break;
case DevFmtUShort:
Write_ALushort(device, buffer, SamplesToDo);
break;
case DevFmtFloat:
Write_ALfloat(device, buffer, SamplesToDo);
break;
}
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}
size -= SamplesToDo;
}
RestoreFPUMode(fpuState);
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}
ALvoid aluHandleDisconnect(ALCdevice *device)
{
ALCcontext *Context;
LockDevice(device);
device->Connected = ALC_FALSE;
Context = device->ContextList;
while(Context)
{
ALsource **src, **src_end;
src = Context->ActiveSources;
src_end = src + Context->ActiveSourceCount;
while(src != src_end)
{
if((*src)->state == AL_PLAYING)
{
(*src)->state = AL_STOPPED;
(*src)->BuffersPlayed = (*src)->BuffersInQueue;
(*src)->position = 0;
(*src)->position_fraction = 0;
}
src++;
}
Context->ActiveSourceCount = 0;
Context = Context->next;
}
UnlockDevice(device);
}